Digital signal processing apparatus, digital signal processing method and data recording medium

ABSTRACT

This invention employs a scheme to extract, from components within a plurality of blocks obtained by subdividing an input signal with respect to time and frequency, every respective blocks, a component or plural components in order of magnitude of components within the respective blocks to determine, on the basis of a difference between magnitudes of components of respective blocks except for the extracted components and magnitudes of the extracted components, a bit allocation ratio to the respective blocks to quantize components of respective blocks on the basis of the bit allocation ratio thus to generate compressed data, thereby making it possible to realize a technique of allocation of bits desirable also from a viewpoint of the auditory sense with respect to such an input signal including, e.g., overtone to much degree. Accordingly, it is possible to carry out efficient compression/expansion of high sound quality from a viewpoint of the hearing sense. In addition, data recording medium adapted for recording therein data compressed by the digital signal processing apparatus of this invention can more effectively utilize memory capacity as compared to the conventional data recording medium.

DESCRIPTION

1. Technical Field

This invention relates to a recording/reproduction of compressed data inwhich digital audio signals, etc. are caused to undergo bit compression,a recording medium adapted so that those compressed data are recordedtherein, and a transmission system for compressed data, and moreparticularly to a digital signal processing apparatus, a digital signalprocessing method and a data recording medium, which are suitable whenused in the case of carrying out, in dependency upon amplitude change ofa waveform on the time base of an input signal, information compressionof such a digital signal to vary size in point of time (length) of ablock subject to processing thereof to conduct recording or transmissionand/or reproduction or reception of such a compressed digital signal toexpand it.

2. Background Art

The applicant of this application has already proposed, e.g., in theJapanese Patent Application Laid Open No. 105269/1992, the JapanesePatent Application Laid Open No. 105270/1992, the Japanese PatentApplication Laid Open No. 105271/1992, and the Japanese PatentApplication Laid Open No. 6572/1993 of the Japanese Laid Open PatentPublications, etc., such a technology to implement bit compression to aninputted digital audio signal to record them in a burst manner with apredetermined data quantity being as a recording unit.

This technology is a technology using a magneto-optical disc as arecording medium and adapted for recording/reproducing AD (AdaptiveDifference) PCM audio data standardized by the audio data format of socalled CD-I (CD-Interactive), CD-ROM XA, wherein such audio data arerecorded onto the magneto-optical disc in a burst manner with, e.g., 32sectors of the ADPCM data and several linking sectors for interleavingprocessing being as a recording unit.

In the case of such ADPCM audio data in a recording/reproducingapparatus using a magneto-optical disc, several modes can be selected.For example, there are prescribed (provided), e.g., level A wherecompression rate is 2 when viewed from comparison with reproduction timeof the ordinary CD and sampling frequency is 37.8 kHz, level B wherecompression rate is 4 when similarly viewed and sampling frequency is37.8 kHz, and level C where compression rate is 8 when similarly viewedand sampling frequency is 18.9 kHz. Namely, for example, in the case ofthe level B, digital audio data is compressed into substantially onefourth (1/4), and reproduction time (play time) of the disc recorded inthe mode of the level B becomes equal to a value four times greater thanthat in the case of the standard CD format (CD-DA format). This permitsthe apparatus to become compact because recording/reproducing time whichis the same order as that of the more compact standard 12 cm disc can beobtained.

In this instance, since rotation speed of the disc is the same as thatof the standard CD, compressed data corresponding to reproduction timefour times greater than that of the standard CD per a predetermined timewill be obtained, e.g., in the case of the level B. For this reason, anapproach is employed to read out four times, in a overlapping manner,the same compressed data in time units, e.g., sector or cluster, etc. touse, for reproduction of audio data, only data of one read-out operationthereof. In actual terms, in scanning (tracking) recording tracks in aspiral form, such a track jump to return to the original track positionevery one rotation is carried out to successively conduct reproducingoperation in such a form to carry out tracking of the same trackrepeatedly by four times. This means that it is sufficient that normal(correct) compressed data can be obtained by at least only one readoperation of, e.g., four times of overlapping (repetitive) readoperations. Employment of such an approach is tolerant of error bydisturbance, etc. and is preferable particularly when applied toportable compact (small-sized) equipments.

Further, the applicant of this application has proposed, in the JapanesePatent Application Laid Open No. 206866/1993 of the Japanese Laid OpenPatent Publication, a bit allocation technique for efficiently realizingsatisfactory compression. In this technology, in allocation of bits, bitallocation dependent upon magnitudes of signals within respective smallblocks such as so called critical bands, etc. is carried out whileimplementing weighting in accordance with corresponding bands of thesmall blocks. According to this technology, in the case where extremeunevenness does not take place in magnitude of spectrum componentswithin respective small blocks, compression can be satisfactorilycarried out.

However, in the case where extreme unevenness or conspicuous peakcomponent is included in magnitudes of spectrum components withinrespective small blocks, in other words, in the case where sound to bemasked is tone-shaped, or in the case of small blocks where there is noextreme unevenness in magnitudes of signals (signal components) withinrespective small blocks indicating values similar to values whenmagnitudes of signals within respective small blocks are determined byvalues which take the maximum values within respective small blocks orsum total values or mean values of magnitudes of signals withinrespective small blocks, sound to be masked can not be distinct from anoise-shaped signal within small block. Thus, there may occur instanceswhere satisfactory result cannot be obtained even with theabove-described technology. This results from the fact that when themasking effect is taken into consideration, degree of the effect variesin dependency upon the property of sound to be masked, i.e., whetherthat sound is noise-shaped or tone-shaped.

Accordingly, with the above-described technology, in the above-mentionedcase, a greater number of bits are required, i.e., bit allocation causedto be in correspondence with a small block where a tone-shaped signal issound to be masked must be employed. As a result, excess bits would beallocated to a small block where a masked sound is noise-shaped, inwhich it is sufficient to implement allocation of a lesser number ofbits is noise-shaped, giving rise to instances where efficiency incompression may be lowered.

Thus, this invention has been made in view of such actual circumstances,and its object is to provide a digital signal processing apparatus, etc.to which a technique of bit allocation caused to be in correspondencewith the property of sound to be masked is applied.

DISCLOSURE OF THE INVENTION

This invention has been proposed in order to attain the above-describedobject, and a first digital signal processing apparatus according tothis invention is directed to a digital signal processing apparatusadapted for compressing a digital signal to record or transmit it,characterized in that the apparatus comprises: extracting means forextracting, from components within a plurality of blocks obtained bysubdividing an input signal with respect to time and frequency, everyrespective blocks, a component or plural components in order ofmagnitude of the components within each of the blocks; bit allocatingmeans for determining, on the basis of a difference between magnitudesof components of the respective blocks except for the extractedcomponents and magnitudes of the extracted components, a bit allocationratio to the respective blocks; and encoding means for quantizingcomponents of the respective blocks on the basis of the bit allocationratio to generate compressed data.

Moreover, a second digital signal processing apparatus according to thisinvention is characterized in that, in the first digital signalprocessing apparatus, the encoding means normalizes components withinthe respective blocks by representative values within the respectiveblocks.

Further, a third digital signal processing apparatus according to thisinvention is characterized in that, in the first digital signalprocessing apparatus, the bit allocating means further determines thebit allocation ratio on the basis of magnitudes of components within therespective blocks and so that weighting is carried out in accordancewith corresponding bands of the respective blocks.

In addition, a fourth digital signal processing apparatus according tothis invention is characterized in that, in the first digital signalprocessing apparatus, the extracting means switches the number of theextracted components in accordance with corresponding bands of therespective blocks.

A first digital signal processing method according to this invention isdirected to a digital signal processing method of compressing a digitalsignal to record or transmit it, characterized in that the methodcomprising the steps of: extracting, from components within a pluralityof blocks obtained by subdividing an input signal with respect to timeand frequency, every respective blocks, a component or plural componentsin order of magnitude of components within each of the blocks;determining, on the basis of a difference between magnitudes ofcomponents of the respective blocks except for the extracted componentsand magnitudes of the extracted components, a bit allocation ratio tothe respective blocks; and quantizing components of the respectiveblocks on the basis of the bit allocation ratio to generate compresseddata.

Moreover, a second digital signal processing method according to thisinvention is characterized in that, in the first digital signalprocessing method, the method further includes a step of normalizingcomponents within the respective blocks by representative values withinthe respective blocks.

Further, a third digital signal processing method according to thisinvention is characterized in that, in the first signal processingmethod, the method comprises a step of determining the bit allocationratio on the basis of magnitudes of components of the respective blocksand so that weighting is carried out in accordance with correspondingbands of the respective blocks.

In addition, a fourth signal processing method according to thisinvention is characterized in that, in the first digital signalprocessing method, the number of the extracted components is switched inaccordance with corresponding bands of the respective blocks.

A first data recording medium according to this invention is directed toa data recording medium adapted so that compressed data are recordedtherein, characterized in that the data recording medium is formed bythe steps of: extracting, from components within a plurality of blocksobtained by subdividing an input signal with respect to time andfrequency, every respective blocks, a component or plural components inorder of magnitude of components within each of the blocks; determininga bit allocation ratio to the respective blocks on the basis of adifference between magnitudes of components of the respective blocksexcept for the extracted components and magnitudes of the extractedcomponents; quantizing components of the respective blocks on the basisof the bit allocation ratio to generate compressed data; and recordingthe compressed data onto or into the recording medium.

Moreover, a second data recording medium according to this invention ischaracterized in that, in the first data recording medium, the datarecording medium is formed by further including a step of normalizingcomponents within the respective blocks by representative values withinthe respective blocks.

Further, a third data recording medium according to this invention ischaracterized in that, in the first data recording medium, the datarecording medium is formed by further including a step of determiningthe bit allocation ratio on the basis of magnitudes of components of therespective blocks and so that weighting is carried out in accordancewith corresponding bands of the respective blocks.

In addition, a fourth data recording medium according to this inventionis characterized in that, in the first data recording medium, the datarecording medium is formed by further including a step of switching thenumber of the extracted components.

Namely, this invention employs a scheme to extract, from componentswithin a plurality of blocks obtained by subdividing an input signalwith respect to time and frequency, every respective blocks, a componentor components in order of magnitude of the components within each of theblocks to determine a bit allocation ratio to the respective blocks onthe basis of a difference between magnitudes of components of therespective blocks except for the extracted components and magnitudes ofthe extracted components to quantize components of the respective blockson the basis of the bit allocation ratio to generate compressed data,thereby making it possible to prevent lowering of efficiency ofcompression. Thus, it becomes possible to obtain a more satisfactorysound quality at the same bit rate, and it becomes possible to realizerecording/transmission, etc. at a lower bit rate with respect to thesame sound quality.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block circuit diagram showing an example of theconfiguration of a recording/reproducing apparatus (discrecording/reproducing apparatus) for compressed data as an embodiment ofa digital signal processing apparatus according to this invention.

FIG. 2 is a block circuit diagram showing an actual example of anefficient compression encoding encoder which can be used for bit ratecompression encoding of this embodiment.

FIGS. 3(A)-3(D) are views showing the structure of orthogonal transformblock in bit compression.

FIG. 4 is a block circuit diagram showing an example of theconfiguration of a circuit for determining orthogonal transform blocksize.

FIGS. 5(A)-5(C) are views showing the relationship between change oflength in point of time of orthogonal transform blocks adjacent in pointof time and window shape used at the time of orthogonal transformprocessing.

FIG. 6 is a view showing an example of the detail of shape of windowused at the time of orthogonal transform processing.

FIG. 7 is a block circuit diagram showing an example of an adaptive bitallocation circuit for realizing bit allocation operation (calculating)function.

FIGS. 8(A)-8(B) are views showing effect of peak component dependent bitallocation.

FIGS. 9(A)-9(B) are views showing bit allocation when a signal having arelatively flat and noise shaped spectrum (waveform) is inputted.

FIGS. 10(A)-10(B) are views showing bit allocation when a signal havingpeak component is inputted.

FIG. 11 is a block circuit diagram showing an actual example of anefficient compression encoding decoder which can be used for bit ratecompression encoding of the embodiment.

BEST MODE FOR CARRYING OUT THE INVENTION

Initially, FIG. 1 is a block circuit diagram showing outline of theconfiguration of an embodiment of a digital signal processing apparatus(compressed data recording and/or reproducing apparatus 9) of thisinvention.

In the compressed data recording and/or reproducing apparatus 9 shown inFIG. 1, a magneto-optical disc 1 rotationally driven by a spindle motor51 is used as a recording medium. At the time of recording data withrespect to the magneto-optical disc 1, e.g., a modulation magnetic fieldcorresponding to recording data is applied by means of a magnetic head54 under the state where laser beams are irradiated by an optical head53 to thereby carry out so called magnetic field modulation recording torecord data along recording tracks of the magneto-optical disc 1. On theother hand, at the time of reproduction, recording tracks of themagneto-optical disc i is traced by laser beams by means of the opticalhead 53 to magneto-optically carry out reproduction.

The optical head 53 comprises, e.g., laser light source such as laserdiode, etc., collimator lens, objective (object lens), polarizing beamsplitter, optical parts such as cylindrical lens, etc., andphotodetector having a light receiving section of a predeterminedpattern, etc. This optical head 53 is provided at the position oppositeto the magnetic head 54 through the magneto-optical disc 1. At the timeof recording data onto the magneto-optical disc 1, the magnetic head 54is driven by a head drive circuit 66 of the recording system which willbe described later to apply modulation magnetic field corresponding torecording data, and to irradiate laser beams onto the target track ofthe magneto-optical disc i by means of the optical head 53 to therebycarry out thermal magnetic recording by the magnetic field modulationsystem. Moreover, this optical head 53 detects a reflected light oflaser beams irradiated onto the target track to detect focus error,e.g., by so called astigmatism method, and to detect tracking error,e.g., by so called push-pull method. At the time of reproducing datafrom the magneto-optical disc 1, the optical head 53 detects focus erroror tracking error as described above, and detects a difference ofpolarization angle (Kerr rotational angle) of a reflected light from thetarget track of laser beams to generate a reproduction signal.

An output of the optical head 53 is delivered to a RF circuit 55. ThisRF circuit 55 extracts the focus error signal or the tracking errorsignal from the output of the optical head 53 to deliver it to a servocontrol circuit 56, and to binarize the reproduction signal (allow thereproduction signal to be a binary signal) to deliver it to a decoder 71of the reproducing system which will be described later.

The servo control circuit 56 comprises, e.g., focus servo controlcircuit, tracking servo control circuit, spindle motor servo controlcircuit, and sled servo control circuit, etc. The focus servo controlcircuit carries out focus control of the optical system of the opticalhead 53 so that the focus error signal becomes equal to zero. Moreover,the tracking servo control circuit carries out tracking control of theoptical system of the optical head 53 so that the tracking error signalbecomes equal to zero. Further, the spindle motor servo control circuitcontrols the spindle motor 51 so as to rotationally drive themagneto-optical disc 1 at a predetermined rotation velocity (e.g.,constant linear velocity). In addition, the sled servo control circuitmoves the optical head 53 and the magnetic head 54 to the target trackposition of the magneto-optical disc 1 designated by a system controller57. The servo control circuit 56 serving to carry out various controloperations as described above sends, to the system controller 57,information indicating operating states of respective componentscontrolled by the servo control circuit

A key input operation section 58 and a display section 59 are connectedto the system controller 57. This system controller 57 carries outcontrol of the recording system and the reproducing system in anoperation mode designated by operation input information from the keyinput operation section 58. Moreover, the system controller 57 carriesout, on the basis of address information of sector unit reproduced byheader time or Q data of subcode, etc. from a recording track of themagneto-optical disc 1, management of recording position or reproductionposition on the recording track that the optical head 53 and themagnetic head 54 are tracing. Further, the system controller 57 carriesout, on the basis of data compression rate and reproduction positioninformation on the recording track, a control to allow the displaysection 59 to display reproduction time.

With respect to the reproduction time display, address information(absolute time information) of sector unit reproduced by so calledheader time, or so called subcode Q data, etc. from recording tracks ofthe magneto-optical disc 1 is multiplied by inverse number of datacompression rate (e.g., 4 in the case of 1/4 compression) to therebydetermine actual time information to allow the display section 59 todisplay it. It is to be noted that, also at the time of recording, forexample, in the case where absolute time information is recorded inadvance (pre-formatted) on recording tracks of magneto-optical disc,etc., an approach may be employed to read the pre-formatted absolutetime information to multiply it by inverse number of data compressionrate to thereby permit a current position to be displayed by an actualrecording time.

In the recording system of the recording/reproducing section of the discrecording/reproducing apparatus, an analog audio input signal AIN froman input terminal 60 is delivered to an A/D converter 62 through alow-pass filter 61. This A/D converter 62 quantizes the analog audioinput signal AIN. A digital audio signal obtained from the A/D converter62 is delivered to an ATC (Adaptive Transform Coding) PCM encoder 63.Moreover, a digital audio input signal DIN from an input terminal 67 isdelivered to the ATC encoder 63 through a digital input interfacecircuit 68. The ATC encoder 63 carries out bit compression (datacompression) processing with respect to digital audio PCM data of apredetermined transfer rate (speed) obtained by quantizing the inputsignal AIN by means of the A/D converter 62. While explanation will nowbe given on the assumption that the compression rate is 4(magnification), this embodiment employs a configuration which is notdependent on the above-mentioned magnification, and the magnificationmay be arbitrarily selected depending upon applications.

A memory 64 is adapted so that write and read operations of data arecontrolled by the system controller 57, and is used as a buffer memoryfor temporarily storing ATC data delivered from the ATC encoder 63 torecord it onto the disc as occasion demands. Namely, with respect toe.g., compressed audio data delivered from the ATC encoder 63, its datatransfer rate (speed) is reduced to one fourth (1/4) of data transferrate (75 sectors/sec.) of the standard CD-DA format, i.e., 18.75sectors/sec. Such compressed data are successively written into thememory 64. For such compressed data (ATC) data, it is sufficient tocarry out recording of one sector per four sectors as previouslydescribed. However, since recording every four sectors (i.e., everyother three sectors) is almost unable to be carried out from a practicalpoint of view, sector continuation recording which will be describedlater is carried out. This recording is carried out in a burst manner ata data transfer rate (speed) (75 sectors/sec.) which is the same as thatof the standard CD-DA format, with cluster comprised of predeterminedplural sectors (e.g., 32 sectors+several sectors) being as a recordingunit, through idle (interruptive) time period. Namely, from the memory84, ATC audio data which have been successively written at a lowtransfer rate of 18.75 (=75/4) sectors/sec. corresponding to the bitcompression rate are read out in a burst manner at the 75 sectors/sec.as recording data. While, with respect to the data which are read outand recorded, the entire data transfer rate including the recording idle(interruptive) time period is the low rate of 18.75 sectors/sec.,instantaneous data transfer rate within time of the recording operationcarried out in a burst manner is the standard 75 sectors/sec.Accordingly, when the disc rotation velocity is the same velocity as thestandard CD-DA format (constant linear velocity), recording having thesame recording density and memory pattern as those of the CD-DA formatwill be carried out.

ATC audio data, i.e., recording data which has been read out in a burstmanner at the (instantaneous) transfer rate of 75 sectors/sec. from thememory 64 is delivered to an encoder 65. In this instance, in the datatrain delivered from the memory 64 to the encoder 65, unit where dataare continuously recorded by single recording is caused to be a clustercomprised of a plurality of sectors (e.g., 32 sectors) and severalsectors for cluster connection allocated at positions before and afterthe cluster. The length of the cluster connection sector is set to avalue longer than the interleaving length at the encoder 65 so that evenif data is interleaved, this does not affect data of other clusters.

The encoder 65 implements encoding processing for error correction(addition of parity and interleaving processing) or EFM encodingprocessing, etc. to the recording data delivered in a burst manner asdescribed above from the memory 64. The recording data to which encodingprocessing by the encoder 65 has been implemented is delivered to themagnetic head drive circuit 66. To this magnetic head drive circuit 66,the magnetic head 54 is connected. Thus, the magnetic head drive circuit66 drives the magnetic head 54 so as to apply modulation magnetic fieldcorresponding to the recording data onto the magneto-optical disc 1.

Moreover, the system controller 57 carries out a memory control asdescribed above with respect to the memory 64, and carries out controlof recording position so as to successively record, onto recordingtracks of the magneto-optical disc 1, the recording data read out in aburst manner from the memory 64 by the above-mentioned memory control.Such control of recording position is carried out by conductingmanagement of recording position of the recording data read out in aburst manner from the memory 64 by the system controller 57 to deliver,to the servo control circuit 56, a signal to designate a recordingposition on the recording track of the magneto-optical disc 1.

The reproducing system of the magneto-optical disc recording/reproducingunit will now be described. This reproducing system is a system forreproducing recording data continuously recorded on recording tracks ofthe magneto-optical disc 1 by the above-described recording system, andincludes a decoder 71 supplied with a reproduction output, which hasbeen binarized by the RF circuit 55, obtained by tracing recordingtracks of the magneto-optical disc 1 by laser beams by means of theoptical head 53. At this time, it is possible to carry out read-outoperation of not only the magneto-optical disc but also a reproductiononly optical disc which is the same as compact disc (CD).

The decoder 71 corresponds to the encoder 65 in the above-describedrecording system, and serves to carry out, with respect to the binarized(binary) reproduction output by the RF circuit 55, processing such asdecoding processing or EFM decoding processing as described above forerror correction to reproduce audio data at a transfer rate (speed) of75 sectors/sec. which is higher than the normal transfer rate (speed).The reproduction data obtained by the decoder 71 is delivered to amemory 72.

The memory 72 is adapted so that write and read operations of data arecontrolled by the system controller 57. Reproduction data delivered atthe transfer rate of 75 sectors/sec. from the decoder 71 is written in aburst manner into the memory 72 at that transfer rate of 75 sectors/sec.Moreover, from the memory 72, the reproduction data which has beenwritten in burst manner at the transfer rate of 75 sectors/sec. arecontinuously read out at a transfer rate of 18.75 sectors/sec. which isone fourth (1/4) of the normal 75 sectors/sec. transfer rate.

The system controller 57 carries out such a memory control to writereproduction data into the memory 72 at the transfer rate of 75sectors/sec., and to continuously read out the reproduction data at thetransfer of 18.75 sectors/sec. from the memory 72. Moreover, the systemcontroller 57 carries out a memory control as described above withrespect to the memory 72, and carries out control of reproductionposition so as to continuously reproduce, from recording tracks of themagneto-optical disc 1, the reproduction data read out in burst mannerfrom the memory 72 by that memory control. Such control of reproductionposition is conducted by carrying out management of reproductionposition of the reproduction data read out in burst manner from thememory 72 by the system controller 57 to deliver, to the servo controlcircuit 56, a control signal to designate a reproduction position onrecording tracks of the magneto-optical disc 1 or the optical disc 1.

ATC audio data obtained as reproduction data which has been continuouslyread out at the transfer rate of 18.75 sectors/sec. from the memory 72is delivered to an ATC decoder 73. This ATC decoder 73 implements dataexpansion (bit expansion) to the ATC data so that the data quantitybecomes quadruple to thereby reproduce digital audio data of 16 bits.The digital audio data from the ATC decoder 73 is delivered to a D/Aconverter

The D/A converter 74 converts the digital audio data delivered from theATC decoder 73 into an analog signal thus to form an analog audio outputsignal A OUT. The analog audio signal A OUT obtained by the D/Aconverter 74 is outputted from output terminal 76 through a low-passfilter

Efficient compression encoding in the ATC encoder 63 will now bedescribed in detail. Namely, explanation will be given with reference toFIG. 2 and figures succeeding thereto in connection with the technologyfor efficiently encoding an input digital signal such as an audio PCMsignal, etc. by using respective technologies Sub Band Coding (SBC),Adaptive Transform Coding (ATC) and adaptive bit allocation.

In the more practical efficient encoding unit (apparatus) shown in FIG.2, an approach is employed to divide an input digital signal into signalcomponents in a plurality of frequency bands, and to make a selection(setting) such that bandwidths of adjacent two bands of the lowestfrequency band are equal to each other and bandwidths become broader inhigher frequency bands according as frequency shifts to higher frequencyband side to carry out orthogonal transform processing every respectivefrequency bands to adaptively carry out bit allocation of the spectrumdata of the frequency base (axis) thus obtained, in lower frequencybands, every so called critical bands in which the hearing sensecharacteristic of the human being is taken into consideration, whichwill be described later and, in medium and higher frequency bands, everybands obtained by subdividing the critical bands width by taking blockfloating efficiency into consideration, thus to encode those data.Ordinarily, such blocks become quantizing noise generation blocks.Further, in the embodiment of this invention, prior to orthogonaltransform processing, block sizes (block lengths) are adaptively variedin dependency upon an input signal, and floating processing are carriedout in the block units.

Namely, in FIG. 2, input terminal 200 is supplied with an audio PCMsignal of 0˜22 kHz when sampling frequency is, e.g., 44.1 kHz. Thisinput signal is divided into a signal in 0˜11 kHz band and a signal in11 kHz˜22 kHz band by a band division filter 201, e.g., so called QMFfilter, etc. The signal in the 0˜11 kHz band is divided into a signal ina 0 kHz˜5.5 kHz band and a signal in a 5.5 kHz˜11 kHz band similarly bya band division filter 202 such as so called QMF filter, etc. The signalin the 11 kHz˜22 kHz band from the band division filter 201 is sent to aMDCT circuit 203 which is an example of the orthogonal transformcircuit, the signal in the 5.5 kHz˜11 kHz band from the band divisionfilter 202 is sent to a MDCT circuit 204, and the signal in the 0˜5.5kHz band from the band division filter 202 is sent to a MDCT circuit205. Thus, those signals are respectively caused to undergo MDCTprocessing.

As a technique for dividing the above-described input digital signalinto signals (signal components) in a plurality of frequency bands,there is, e.g., QMF filter, which is described in 1976 R. E. CrochiereDigital Coding of Speech In Subbands Bell Syst. Tech. J. Vol. 55, No. 81976. Moreover, a filter division technique of equal bandwidth isdescribed in ICASSP 83, Boston Polyphase Quadrature Filters--A NewSubband Coding Technique Joseph H. Rothweiler.

Further, as the above-described orthogonal transform processing, thereis, e.g., such an orthogonal transform processing to divide an inputaudio signal into blocks every predetermined unit time (frame) to carryout, every respective blocks, Fast Fourier Transform (FFT), DiscreteCosine Transform (DCT), or Modified DCT (MDCT), etc. to transformsignals on the time axis into signals on the frequency axis. The MDCTmentioned above is described in ICASSP 1987 Subband/Transform CodingUsing Filter Designs Based On Time Domain Aliasing Cancellation J. P.Princen A. B. Bradley Univ. of Surrey Royal Melbourne Inst. Of Tech.

An actual example with respect to the standard input signal inconnection with blocks every respective bands, which are delivered torespective MDCT circuits 203, 204, 205, is shown in FIG. 3. In theactual example of FIG. 3, three filter output signals respectivelyindependently have, every respective bands, plural orthogonal transformblock sizes, and time resolution can be switched by time characteristicand frequency distribution of signal, etc. In the case where a signal isin quasi-steady state, orthogonal transform block size is set to 11.6ms, i.e., Long Mode of (A) in FIG. 3 is employed so that block size isgreater (longer). On the other hand, in the case where a signal is innon steady state, orthogonal transform block size is further dividedinto two or four portions. In this case, an approach is employed inwhich block size is divided into four portions so that each has timeresolution of 2.9 ms as in the case of Short Mode of (B) in FIG. 3, orone portion obtained by dividing block size into two portions so that ithas time resolution of 5.8 ms and portions obtained by dividing blocksize into four portions so that they have time resolution of 2.9 ms asin the case of Middle Mode A of (C), Middle Mode B of (D) in FIG. 3,thus to become adaptive to actual complicated input signals. It is clearthat with respect to division of the orthogonal transform block size, ifscale of the processing unit is tolerable, implementation of furthercomplicated division is more effective. Determination of the block sizeis carried out at block size determining circuits 206, 207, 208 in FIG.2. The block sizes thus determined are sent to respective MDCT circuits203, 204, 205, and are outputted from output terminals 216, 217, 218 asblock size information of the corresponding blocks.

The detail of the block size determining circuit is shown in FIG. 4.Explanation will be given by taking an example of block determiningcircuit 206 in FIG. 2. An output of 11 kHz˜22 kHz of outputs of QMF 201is sent to a power calculating circuit 404 through an input terminal 401in FIG. 4. Further, an output of 5.5 kHz˜11 kHz of outputs of QMF 202 inFIG. 2 is sent to a power calculating circuit 405 through an inputterminal 402 in FIG. 4, and an output of 0˜5.5 kHz is sent to a powercalculating circuit 406 through an input terminal 403 in FIG. 4. It isto be noted that block size determining circuits 207, 208 in FIG. 2 arethe same in operation as the block size determining circuit 206 exceptthat signals inputted to input terminals 401, 402, 403 in FIG. 4 aredifferent from that in the case of the block size determining circuit206. Respective input terminals 401, 402, 403 in the block sizedetermining circuits 206, 207, 208 are caused to have matrixconfiguration. Namely, output of 5.5 kHz˜11 kHz of QMF 202 in FIG. 2 isconnected (sent) to the input terminal 401 of the block size determiningcircuit 207, and output of 0˜5.5 kHz is connected (sent) to the inputterminal 402 thereof. This similarly applies to the block sizedetermining circuit 208.

In FIG. 4, respective power calculating circuits 404, 405, 406 integratean inputted time waveform for a predetermined time to thereby determine(calculate) powers of respective frequency bands. In this instance, itis necessary that time width subject to integration is less than aminimum time block of the above-described orthogonal transform blocksizes. Moreover, in addition to the above-described calculating method,if e.g., absolute value of the maximum amplitude or mean value ofamplitudes within a minimum time width of the orthogonal transform blocksize is employed as a representative power, similar effect can beobtained. An output of the power calculating circuit 404 is sent to achange (value) extracting circuit 408 and a power comparing circuit 409,and outputs of power calculating circuits 405, 406 are both sent to thepower comparing circuit 409. The change extracting circuit 408determines differential coefficient of the power sent from the powercalculating circuit 404 to send it, as change information of power, to ablock size temporary determining circuit 410 and a memory 407. Thememory 407 stores the power change information sent from the changeextracting circuit 408 for a time period more than the maximum time ofthe above-described orthogonal transform block size. This is becausesince orthogonal transform blocks adjacent in point of time are affectedeach other by window (windowing) processing in orthogonal transformprocessing, power change information of a block preceding (backward) byone adjacent in point of time is required in the block size temporarydetermining circuit 410. The block size temporary determining circuit410 determines, on the basis of power change information of acorresponding block sent from the change extracting circuit 408 andpower change information of a block preceding (backward) by one of thecorresponding block adjacent in point of time sent from the memory 407,an orthogonal block size of the corresponding frequency band from adisplacement (shift) in point of time of a power within thecorresponding frequency band. In this instance, in the case where adisplacement (shift) of a predetermined level (value) or more isobserved, a shorter orthogonal block size in point of time is selected.In this case, even if that point of displacement (shift) is fixed,effect can be obtained. Further, when determination is made such thatdisplacement (shift) point becomes a value proportional to frequency,i.e., in the case where frequency is high, block size is caused to be ablock size shorter in point of time by great displacement (shift), whilein the case where frequency is low, block size is caused to be a blocksize shorter in point of time by displacement (shift) smaller than thatin the case where frequency is high, more effective result is obtained.Although it is desirable that such value smoothly changes, stair-stepshaped change of plural stages may be employed. The block sizedetermined in a manner as described above is sent to a block sizemodifying circuit 411.

On the other hand, the power comparing circuit 409 makes comparisonbetween power information of respective frequency bands sent fromrespective power calculating circuits 404, 405, 406 by time widthgenerated by (based on) the simultaneous masking effect and the maskingeffect on the time base to determine influence of other frequency bandsexerted on an output frequency band of the power calculating circuit 404to send it to the block size modifying circuit 411. The block sizemodifying circuit 411 makes (applies) a modification, on the basis ofmasking information sent from the power comparing circuit 409 and pastblock size information sent from respective taps of a group of delays412, 413, 414, so as to make a selection to allow the block size sentform the block size temporary determining circuit 410 to be a block sizelonger in point of time to output it to the delay 412 and a window shapedetermining circuit 415. Action (operation) in the block size modifyingcircuit 411 utilizes the characteristic that even in the case wherepre-echo becomes problem in a corresponding frequency band, there areinstances where when any signal having a large amplitude exists in otherfrequency bands, particularly frequency bands lower than thecorresponding frequency band, pre-echo does not become problem from aviewpoint of the hearing sense or the problem by pre-echo is lessened bythat masking effect. It should be noted that the masking refers to thephenomenon that a signal is masked by another signal by thecharacteristic from a viewpoint of the hearing sense of the human beingso that it cannot be heard. For such masking effect, there are time axismasking effect by audio signal on the time axis and simultaneous maskingeffect by signal on the frequency axis. By these masking effects, evenif any noise exists at the portion subjected to masking, such noisewould not be heard. For this reason, in actual audio signals, noisewithin the range subjected to masking is considered to be allowablenoise.

The group of delays 412, 413, 414 record past orthogonal transform blocksizes in order to output them to the block size determining circuit 411from respective taps, i.e., outputs (output terminals) of the group ofdelays 412, 413, 414. At the same time, an output (terminal) of thedelay 412 is connected to an output terminal 417, and outputs (outputterminals) of the group of delays are connected to the window shapedetermining circuit 415. Outputs from the group of delays 412, 413, 414permit judgment of how change of block size at a longer time width iscaused to serve to determine block size of a corresponding block in theblock size modifying circuit 411, for example, such that the degree ofselection of block sizes shorter in point of time is increased whenblock sizes short in point of time are frequently selected in the past,and the degree of selection of block sizes longer in point of time isincreased when no selection of block size shorter in point of time ismade in the past, etc. It is to be noted that there are instances where,except for delays 412, 413 required for window shape determining circuit415 and output terminal 417, the group of delays may be used with thenumber of taps thereof being increased or decreased in dependency uponactual configuration or scale of the apparatus. The window shapedetermining circuit 415 determines, from an output of the block sizemodifying circuit 411, i.e., a block size succeeding (forward) by oneadjacent in point of time of a corresponding block, an output of thedelay 412, i.e., block size of the corresponding block, an output of thedelay 413, i.e., a block size preceding (backward) by one adjacent inpoint of time of the corresponding block, shapes of windows used in therespective MDCT circuits 203, 204, 205 in the FIG. 2 mentioned above tooutput them to the output terminal 416. Output terminal 417 in FIG. 4,i.e., block size information and output terminal 416, i.e., window shapeinformation are connected (sent) to respective components as outputs ofthe block size determining circuits 206, 207, 208 in FIG. 2.

Shape of window determined in the window shape determining circuit 415will now be described. The state of adjacent blocks and shape of windowis shown in FIG. 5. As seen from a˜c of FIG. 5, windows used fororthogonal transform processing have portions overlapping with blocksadjacent in point of time as indicated by dotted lines and solid line(s)in the figure. Since there is employed in this embodiment shapeoverlapping up to the center of the adjacent block, shape of windowchanges in dependency upon orthogonal transform size of the adjacentblock.

Detail of the window shape is shown in FIG. 6. In FIG. 6, windowfunctions f(n), g(n+N) are given by functions which satisfy thefollowing formula:

    f(n)×f(L-1-n)=g(n)×g(L-1-n)

    f(n)×f(n)+g(n)×g(n)=1                          (1)

0≦n≦L-1

L in the above-mentioned formula (1) is considered to be a transformblock length as it is if adjacent transform block lengths are the same.However, in the case where adjacent transform block lengths aredifferent, when it is assumed that a shorter transform block length iscaused to be L and a longer transform block length is caused to be K,window functions are given by the following formula (2) in the areawhere windows do not overlap:

    f(n)=g(n)=1 K≦n≦3K/2-L/2

    f(n)=g(n)=0 3K/2+Ln2K                                      (2)

By allowing overlapping portion of window to be as long as possible inthis way, frequency resolution of spectrum in orthogonal transformprocessing is caused to be satisfactory. As is clear from the foregoingdescription, after orthogonal transform sizes of three blocks continuousin point of time are established, shape of window used for orthogonaltransform processing is determined. Accordingly, in this embodiment,there occur differences corresponding one block between blocks of signalinputted from the input terminals 401, 402, 403 in FIG. 4 and blocks ofsignal outputted from the output terminals 416 and 417.

Moreover, even if power calculating circuits 405, 406 and powercomparing circuit 409 in FIG. 4 are omitted, block size determiningcircuits 206, 207, 208 in FIG. 2 may be constituted. Further, there maybe employed a configuration such that shape of window is fixed to ablock size minimum in point of time that orthogonal transform block cantake to thereby the kind thereof to be one so that group of delays 412,413, 414, block size modifying circuit 411 and window shape determiningcircuit in FIG. 4 are omitted. Particularly in applications where delayof processing time is not preferable, there is provided a configurationhaving lesser delay by the above-described omission, and such omissionof circuit configuration is advantageous.

In FIG. 2, for a second time, spectrum data or MDCT coefficient dataobtained after undergone MDCT processing at respective MDCT circuits203, 204, 205 are sent to adaptive bit allocation encoding circuits 210,211, 213 and a bit allocation calculating circuit 209 in such a mannerthat, in lower frequency bands, those data are combined every blockfloating units in which so called critical bands are caused to be unitwith respect to the frequency axis direction and the block size iscaused be unit with respect to the time axis direction; and in mediumand higher frequency bands, those data are combined every block floatingunits in which bands obtained by subdividing critical bandwidth arecaused to be unit with respect to the frequency axis direction and theblock size is caused to be unit with respect to the time axis direction.Such critical bands are frequency bands divided by taking the hearingsense characteristic of the human being, and are defined as bands thatnarrow band noises having the same intensity in the vicinity offrequency of a certain pure sound have when the pure sound is masked bythose band noises. Such critical bands are such that according asfrequency shifts to higher frequency bands, bandwidths become broader,and the entire frequency band of 0˜22 kHz are divided into, e.g., 25critical bands.

The bit allocation calculating circuit 209 determines, on the basis ofspectrum data divided into the block floating units, masking quantitiesof blocking floating units by taking so called masking, etc. intoconsideration to determine, on the basis of the masking quantities andenergies or peaks, etc. of block floating units, allocated bitallocation ratios every respective block floating units to send them tothe adaptive bit allocation encoding circuits 210, 211, 212. Theseadaptive bit allocation encoding circuits 210, 211, 212 carry outnormalization by using scale factors (e.g., maximum value of absolutevalues of respective components within a corresponding unit) everyrespective block floating units, and calculates, from bit allocationratios allocated every respective block floating units and total numberof usable bits, the numbers of bits which can be actually allocated torespective block floating units, thus to quantize respective spectrumdata (or MDCT coefficient data) in accordance with the calculatednumbers of bits. Data encoded in this way are taken out through outputterminals 213, 214, 215. In this case, the scale factor and a wordlength indicating quantization bit number are also outputted through theoutput terminals 213, 214, 215.

The operation of the bit allocation calculating circuit 209 which is theimportant point of this invention will now be described with referenceto FIG. 7. Respective outputs of MDCT circuit 203, 204, 205 in FIG. 2are connected (sent) to input terminal 700 in FIG. 7, and are inputtedto a peak component extracting circuit 701 and a circuit 702 forcalculating energy every band. The peak component extracting circuit 701implements sequencing to MDCT coefficients within respective blockfloating units so that they are allocated in order of magnitude ofabsolute values of those MDCT coefficients to extract components (alsoincluding one component) in order of magnitude in dependency uponbandwidth of a corresponding block floating unit to output, to asubtracter 703 and a difference calculating element (circuit) 704, avalue (energy of peak component) obtained by square sum of extractedcomponents by total number of all frequency components within thecorresponding block floating unit.

Moreover, the circuit 702 for calculating energy every band determinesmean square of MDCT coefficients every block floating unit to therebycalculate an energy within a corresponding block floating unit to outputit to the subtracter 703. In this instance, if simple mean is determinedin place of mean square, similar effect can be obtained. (At this time,it should be noted that output of the peak component extracting circuit701 is also required to be a value obtained by dividing sum of extractedcomponents by total number of all frequency components within thecorresponding block floating unit.)

Then, the subtracter 703 subtracts outputs of the peak componentextracting circuit 701 from an output of the circuit 702 for calculatingenergy every band to output it to the difference calculating circuit704. Namely, by this calculation, energies except for peak components ofrespective block floating units are calculated.

The difference calculating circuit 704 calculates a difference (at eachtime point) between energy except for the peak component(s) and theenergy of the peak component(s) to output it to a peak componentdependent bit allocation determining circuit 705. While, in thisembodiment, calculation of difference is performed by carrying out inadvance conversion into integer of decimal number (conversion into ID)by the logarithmic axis to calculate differences by ID, it is clear thatif calculation is performed by real number, similar effect can beobtained.

The peak component dependent bit allocation circuit 705 determines bitallocation ratio dependent upon peak component on the basis ofdifference data outputted from the difference calculating circuit 704.In this embodiment, there are prepared a plurality of patterns dependingupon frequency, frequency width of block floating unit and energy ofpeak, e.g., patterns such that a greater number of bits are allocated toa signal of a greater magnitude to carry out determination of bitallocation ratio by table.

The effect of bit allocation dependent upon peak component will now bedescribed with reference to FIG. 8.

When consideration is made in connection with allocation of bitsdependent upon power or energy within the block floating unit, in thecase where attention is drawn to power or energy of peak within theblock floating unit, (a), (b) in FIG. 8 are judged to have the samepower or energy, so substantially the same bits are allocated. On theother hand, when attention is drawn to total energy or mean energywithin the block floating unit, a greater number of bits are allocatedin the case of (a) in FIG. 8. However, as compared to the case of (b) inFIG. 8, in the case of (a) in FIG. 8, power spectrum is clearlynoise-shaped. Therefore, masking effect in the case of (a) is consideredto be higher. Accordingly, there occur instances where employment of bitallocation in which a greater number of bits are rather allocated in thecase of (b) in FIG. 8 provides satisfactory result. In this embodiment,ΔP in FIG. 8 (corresponding to output of the difference calculatingcircuit 704) is calculated to allow it to be included in the bitallocation to thereby obtain satisfactory result. This is based on anauditory sense characteristic such that while the range where themasking effect is exerted in the case where sound to be masked is puresound and that in the case where sound to be masked is noise are nearlyequal to each other, the masking effect in the case of noise is higherbecause beat by two pure sounds does not take place.

In FIG. 7, for a second time, in a fixed bit allocation determiningcircuit 707, a fixed bit allocation ratio is determined from an outputof the circuit 702 for calculating energy every band and a fixed bitallocation pattern table 708. For the fixed bit allocation pattern table708 for determining fixed bit allocation ratio, a plurality of tablesare prepared. Accordingly, it is possible to carry out variousselections in dependency upon the property of signal. In the embodiment,there are provided various patterns in which bit quantities of blocks ofshorter time corresponding to a block subject to processing aredistributed with respect to respective frequencies to select anarbitrary one of them by an output of the circuit 702 for calculatingenergy every band. Particularly, in this embodiment, there are prepareda plurality of patterns in which bit allocation ratio in lower andmedium frequency bands and that in higher frequency band are caused todiffer from each other with respect to a single sum total value ofenergies over the frequency band. Further, there is employed an approachin which according as magnitude of sum total value over the allfrequency bands of output of the circuit 702 for calculating energyevery band becomes smaller, a pattern of a lesser allocated quantity tohigher frequency band is selected. Thus, the loudness effect thataccording as magnitude of a signal becomes smaller, sensitivity of thehigh frequency band is lowered to more degree is exhibited, thus toobtain satisfactory effect. Moreover, while, in this embodiment,selection of fixed bit allocation pattern table 706 is carried out bycalculating energies every bands, an output of non-blocking frequencydividing circuit where filters, etc. are used or MDCT output may beutilized for this purpose.

The fixed bit allocation ratio determined in this way and the peakdependent bit allocation ratio are added by an adder 708. An addedoutput is outputted from output terminal 709 to the adaptive bitallocation encoding circuits 210, 211, 212 in FIG. 2.

The state of the above-described bit allocation is shown in FIGS. 9(b)and 10(b), and the state of MDCT coefficients of an inputted signal withrespect thereto are shown in FIGS. 9(a) and 10(a). In FIGS. 9, 10, forthe brevity of explanation, allocation of bits is represented on theassumption that the entire frequency band is divided into 12 blockfloating units. FIG. 9 shows the case where spectrum of a signal is flatand noise-shaped, and bit allocation by a greater quantity of fixedallocation bits is benefit for taking large signal-to-noise ratios overthe entire bands. In this embodiment, an approach is employed such thatbits are allocated in a form depending upon powers of respective blockfloating units and a greater number of bits are allocated by allocationinclined (weighted) to the lower frequency side. Accordingly, while thesignal noise characteristic at the higher frequency band side isdeteriorated, the signal noise characteristic at the lower frequencyband side is improved. Since noises at the higher frequency sidegenerated by employment of such an approach are difficult to be heard ascompared to noises at the lower frequency side when primarily viewedfrom dependency with respect to frequency of the sensitivity of the earof the human being, and such noises are masked by signal at the lowerfrequency band side, big problem does not result from a viewpoint of thehearing sense.

Further, the state of bit allocation in the case where a signal having aspectrum shown in FIG. 10(a) is inputted is shown in FIG. 10(b), whereinrectangle of white indicates the number of bits allocated to respectiveblocks on the basis of fixed pattern, and rectangle to which slantinglines are implemented indicates the number of bits allocated on thebasis of magnitudes of signal components of respective block floatingunits. The number of bits corresponding to sum of the above-mentionednumbers of bits are allocated to respective block floating units. It isto be noted that while these numeric values are indicated so as to takereal number values in place of integer values, this indicates process ofcalculation, and e.g., an approach may be ultimately employed to roundoff these values to thereby determine allocated bit numbers with respectto respective block floating units. Similarly to the case of FIG. 9,fixed bit allocation is carried out by inclination allocation by powersand frequencies of respective block floating units. Moreover, withrespect to the fifth to the tenth bands, since respective block floatingunits have large peak components therein, allocation of bits dependentupon the peak component is carried out to more degree. In actualsignals, spectrum as described above is observed in a rectangular wavein the case of an artificial signal, and in wave form of windinstruments, etc. in which overtone component is produced withoutattenuating up to relatively higher frequency band.

The noticeable point is that, in the case of the method of thisinvention, while bits lesser than that of the fourth block floating unitare allocated to, e.g., the ninth block floating unit, a greater numberof bits are allocated as compared to the number of bits allocated to,e.g., the eighth block floating unit, etc. Such a bit allocation cannotbe realized by carrying out bit allocation using a technique such asweighting, etc. depending upon magnitudes of signals within respectiveblock floating units and corresponding to frequencies, but can berealized by determining allocation of bits by drawing attention to peakcomponents and other components within respective block floating units.

In this embodiment, it is assumed that, in lower frequency bands lessthan 100 Hz, as spectrum (components) obtained as the result of MDCT,only several spectrum components can be obtained at the maximum. In sucha case, since, in respective spectrum components on the lower frequencyside obtained by calculation, such signals to correspond to frequenciesof bands higher than those lower frequency bands are mixed to muchdegree, it is necessary to allocate sufficiently greater number of bitsto the lower frequency band side. For this reason, speakingapproximately, with respect to bits allocated on the basis of magnitudesof signal components of respective block floating units, many bits maybe allocated according as frequency shifts to lower frequency side.However, if a time period during which spectrum is obtained is caused tobe longer so that an efficient encoding apparatus capable ofsufficiently densely obtaining spectrum less than 100 Hz is employed,allocation of bits dependent upon magnitude of a signal with respect tospectrum corresponding to, e.g., less than 50 Hz where the sensitivityof the ear of the human being is low may be lesser (lower) thanallocation of bits with respect to signal components of bands higherthan that.

In a system as explained above, data obtained by allowing orthogonallytransformed output spectrum to undergo processing (normalization andquantization) is obtained as main information, and scale factorindicating the state of block floating and word length are obtained assub information, and those data are sent from encoder to decoder.

FIG. 11 shows ATC decoder 73 in FIG. 1, i.e., a decoding circuit fordecoding, for a second time a signal which has been caused to undergoefficient encoding. Quantized MDCT coefficients of respective bands,i.e., data equivalent to output signals of output terminals 213, 214,215 in FIG. 2 are delivered to a decoding circuit input 107, and usedblock size information, i.e., data equivalent to output signals ofoutput terminals 216, 217, 218 in FIG. 2 are delivered to an inputterminal 108. An adaptive bit allocation decoding circuit 106 carriesout inverse quantization and normalization by using scale factor andword length. Then, at inverse orthogonal transform (IMDCT) circuits 103,104, 105, signals on the frequency base are transformed into signals onthe time base. The signals on the time base of these partial bands aredecoded into the entire band signal by band synthesis filter (IQMF)circuits 102, 101.

It is to be noted that this invention is not limited to theabove-described embodiment, and, e.g., it is not necessary that theabove-mentioned recording/reproducing medium and the signal compressingor expanding unit, and the signal compressing unit and the signalexpanding unit are integrated, and an approach may be employed toconnect between those units by data transfer line or optical cable, orcommunication by light or radio wave, etc. without allowing a recordingmedium as described above to intervene therebetween. Further, e.g., thisinvention may be applied not only to audio PCM signal but also toprocessing apparatus for signal such as digital speech signal or digitalvideo signal, etc.

Moreover, the data recording medium of this invention records datacompressed by the above-mentioned digital signal processing apparatus,thereby making it possible to effectively utilize recording capacity. Inaddition, as the data recording medium of this invention, not only theabove-described optical disc but also various recording media such asmagnetic disc, IC memory, card including that memory therein, ormagnetic tape, etc. may be employed.

INDUSTRIAL APPLICABILITY

As is clear from the foregoing description, this invention employs ascheme to extract, from components within a plurality of blocks obtainedby subdividing an input signal with respect to time and frequency, everyrespective blocks, a component or plural components in order ofmagnitude of components within each of the blocks to determine, on thebasis of a difference between magnitudes of components of respectiveblocks except for the extracted components and magnitudes of theextracted components, bit allocation ratio to the respective blocks toquantize components of the respective blocks on the basis of the bitallocation ratio, thus to generate compressed data, thereby making itpossible to realize a technique of allocation of bits desirable alsofrom a viewpoint of the auditory sense with respect to such an inputsignal including, e.g., overtone to much degree. Accordingly, it ispossible to carry out efficient compression/expansion of high soundquality from a view point of the hearing sense. In addition, the datarecording medium adapted for recording therein data compressed by thedigital signal processing apparatus of this invention can moreeffectively utilize memory capacity as compared to the conventional datarecording medium.

What is claimed is:
 1. A digital signal processing apparatus adapted forcompressing a digital signal to record or transmit it,the apparatuscomprising: extracting means for extracting, from components within aplurality of blocks obtained by subdividing an input signal with respectto time and frequency, every respective blocks, a component or pluralcomponents in order of magnitude of signal components within each of theblocks; bit allocating means for determining, on the basis of adifference between magnitudes of respective blocks except for theextracted components and magnitudes of the extracted components, a bitallocation ratio to the respective blocks; and encoding means forquantizing components of the respective blocks on the basis of the bitallocation ratio to generate compressed data.
 2. A digital signalprocessing apparatus as set forth in claim 1, wherein the encoding meansnormalizes components within the respective blocks by representativevalues within the respective blocks.
 3. A digital signal processingapparatus as set forth in claim 1, wherein the bit allocating meansfurther determines the bit allocation ratio on the basis of magnitudesof components within the respective blocks and so that weighting iscarried out in accordance with corresponding bands of the respectiveblocks.
 4. A digital signal processing apparatus as set forth in claim1, wherein the extracting means switches the number of the extractedcomponents in accordance with corresponding bands of the respectiveblocks.
 5. A digital signal processing method of compressing a digitalsignal to record or transmit it,the method comprising the steps of:extracting, from components within a plurality of blocks obtained bysubdividing an input signal with respect to time and frequency, everyrespective blocks, a component or plural components in order ofmagnitude of components within the respective blocks; determining, onthe basis of a difference between magnitudes of components of respectiveblocks except for the extracted components and magnitudes of theextracted components, a bit allocation ratio to the respective blocks;and quantizing components of the respective blocks on the basis of thebit allocation ratio to generate compressed data.
 6. A digital signalprocessing method as set forth in claim 5, wherein the method furtherincludes a step of normalizing components within the respective blocksby representative values within the respective blocks.
 7. A digitalsignal processing method as set forth in claim 5, wherein the methodincludes a step of determining the bit allocation ratio on the basis ofmagnitudes of components of the respective blocks and so that weightingis carried out in accordance with corresponding bands of the respectiveblocks.
 8. A signal processing method as set forth in claim 5, whereinthe number of the extracted components is switched in accordance withcorresponding bands of the respective blocks.
 9. A data recording mediumadapted so that compressed data are recorded therein,wherein the datarecording medium is formed by the steps of: extracting, from a pluralityof blocks obtained by subdividing an input signal with respect to timeand frequency, every respective blocks, a component or plural componentsin order of magnitude of components within the respective blocks;determining a bit allocation ratio to the respective blocks on the basisof a difference between magnitudes of components of the respectiveblocks except for the extracted components and magnitudes of theextracted components; quantizing components of the respective blocks onthe basis of the bit allocation ratio to generate compressed data; andrecording the compressed data onto or into the recording medium.
 10. Adata recording medium as set forth in claim 9, wherein the datarecording medium is formed by further including a step of normalizingcomponents within the respective blocks by representative values withinthe respective blocks.
 11. A data recording medium as set forth in claim9, wherein the data recording medium is formed by further including astep of determining the bit allocation ratio on the basis of magnitudesof components of the respective blocks and so that weighting is carriedout in accordance with corresponding bands of the respective blocks. 12.A data recording medium as set forth in claim 9, wherein the datarecording medium is formed by further including a step of switching thenumber of extracted components.